10 h.248,sip,sigtran and service flow of lmds

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Module Code

Module Name

H.248/SIP/SIGTRAN/M3UA/SCTP and Service Flow in LMDS

V0603

ZTE UniversityMobile-MSS/WDSS Group


Understand Function of H.248 ProtocolUnderstand Function of SIP Protocol

Understand Function of SIGTRAN/M3UA/SCTP Protocol

Understand LMSD Service Flow

Course Objectives


Agenda

H.248 ProtocolProtocol Introduction SIP Protocol IntroductionSIP Protocol Introduction SigTran and M3UA / SCTP SigTran and M3UA / SCTP Service Flow in LMSD


What is H.248

MSCe MSCe

MGW

MGW

MGW

Control Signaling

Over IP

H.248H.248

Media

Over IP

H.248 ： used in MSCe to control the MGW and implement

voice bearing on the packet network. They reflect the idea of

separating control function from the media conversion function.

H.248 ： used in MSCe to control the MGW and implement

voice bearing on the packet network. They reflect the idea of

separating control function from the media conversion function.

MGW

MGW


Bearer Mode of H.248 Protocol

IP

SCTP

M3UA

H.248

Upper-layer application


Termination

– Termination is a logical entity on the MG, which initiates/receives media stream. Termination is identified with TerminationID.

MGMG

Ta

Tb

Cn

Tc

Td

I2

O2=I1+I3

I1

I3

O3=I1+I2

O1=I2+I3


There are two types of termination:• Semi-permanent termination Semi-permanent termination represents the physical entity.

For example, for a TDM channel, as long as this channel exists in the MG, the termination exists.

• Temporary termination Temporary termination only exists during the calling

procedure, such as RTP media stream. Once the calling ends, the termination eliminates.

Termination


– Context means the relationship among multiple terminations. If context involves more than two terminations, it describes the topology (receiver/sender), media blending and/or switching parameters.

– When a call is originated, the H.248 protocol can use commands to add terminations corresponding to calling and called parties in the context.

Context


Relationship of H.248 Message/ Transaction / Action /Command

Command1

Command2

Command n

Action 1

Action 2

Action n

Message 1

Message 2

Message n

Transaction 1

Transaction 2

Transaction n

AH


• Command is used to operate and manage logical entities (context and terminations). The protocol defines eight commands, of which the majority is used for the MGC (MSCe) to control MG (MGW).

• All of the commands in one action control the same context.

Commands


H.248 Commands

Add

Modify

Subtract

Move

AuditValue

AuditCapabilities

Notify

ServiceChange

MSCe MGW


Transport

MSCe

MGW

H.248

M3UA/SCTP

IP

Don’t need three-way

Handshake

Port 2944: Text-encoded

Port 2945: Binary-

encoded

H.248


Agenda



SIP, H.323 and H.248

RTCPRTP

IP

H.248/Megaco

Call Control and Signaling Gateway control Media

H.225

Q.931

H.323

TCP

RAS

UDP

SIP

H.245

Video/Audio

RTSP


What is SIP?

SIP: Session Initiation Protocol• SIP is a multimedia communication protocol established

by IETF. It is a text-based application-layer control protocol independent of lower-layer protocols, designed to establish, modify and terminate two-party or multi-party multimedia sessions over the IP network.

“

”


What is SIP? _Cont.

“

”

SIP was firstly researched by the MMUSIC IETF workgroup in 1995 and recommended to be a standard by IETF in 1999.

SIP uses HTTP and SMTP protocols for reference.

SIP is still developing now. Relevant equipment vendors and service providers have created an SIP forum: www.sipforum.org


What is SIP? _Cont.

SIP supports proxy, redirect and user location registration functions. It also supports user mobility. With RTP/RTCP, SDP, RTSP and DNS protocols, it can support and be applied to voice, video and data services. In addition, it can be applied to characteristic services like Presence and Instant Message (similar to QQ).

“

”


Redirect Server

Basic Framework

Location Server

Registrar Server

User Agent

Proxy Server

Gateway

PSTN

SIP Network Members

Proxy Server

SIP

SIP SIP

SIP SIP

LDAP

LDAP


User Agents

– An entity that initiates and terminates sessions. It contains two functional entities.

– User Agent Clients (UAC) –A functional entity that initiates the SIP request.

– User Agent Server (UAS) –A functional entity that receives the SIP request.

– UAC and UAS -Both can terminate a call.

Basic Framework


Basic Framework

Proxy Server– Associated with redirect server and location server

– Proxy for other clients, with SIP message transfer and forwarding functions. The message mechanism of proxy server is similar to that of UAC and UAS.

– Transfer received request messages to other servers after translating and processing.

– Routing the SIP request and response


Basic Framework

Location Server– A database designed to store current location informati

on of the user and used by a SIP REDIRECT or PROXY SERVER to obtain information about a called user’s possible location(s).


Basic Framework

Redirect Server– Associated with location server.

– Return the new location information of a user to the caller. The caller can initiate a new call using the new location information.

– Unlike a proxy server, it does not initiate its own SIP request.

– Unlike a user agent server, it does not accept call termination or terminate a call.


Basic Framework

Registrar Server

– Accept REGISTER request and accomplish user address registration

– Support authentication


Offer SIP users the location function Location mode: SIP URL, similar to the E-mail address format

Performing media format negotiation Encapsulation protocol: SDP - Session Describle Protocol

Initiate a session Request message used: INVITE Change a session Request message used: Re-INVITE (Retransmit INVITE, Cseq is added) Terminate a session Request messages used: BYE, CANCEL

Basic functions of SIP


SIP message - Request and Response

• SIP request:– INVITE- Invite a user to initiate a call. – ACK- Used to acknowledge the final response t

o the INVITE that UA has received, used together with INVITE.

– BYE-USER AGENT uses this method to indicate a call release

– CANCEL- Used to cancel a uncompleted request, which does not affect the completed request.

– REGISTER- The client uses this method to register on the server the addresses in the To field.

– OPTIONS- Used to query the service capabilities.

– INFO- Used to carry outband information, e.g., DTMF information.

• SIP response: – 1xx- Provisional Response.

– 2xx- Successful Response.

– 3xx- Redirection.

– 4xx- Client Error.

– 5xx- Server Error.

– 6xx- Global Failure.

The message mechanism is based on Client/Server, includes Request and Response.


• For Invite, it need three-way Handshake: invite—200 ok—ack.

Bob Laura

(1) INVITE

(2) 180 Ringing

(3) 200 OK

(4) ACK

Sessi on


SIP Session Setup Process

A session will be set up with SIP in the following 6 steps: 1. Register, locate the user.

2. Media negotiation- Media parameters are generally carried in SDP mode.

3. The called party will determine whether this call will be accepted.

4. Media streams are established and interact.

5. Call may be changed or processed, e.g., Call forwarding (CF)

6. Call terminated.


SIP Session Setup and Release

RTP MEDIA PATH

BYE BYE

200 (OK) 200 (OK)Teardown

Session Setup

Proxy Server Proxy Server User AgentUser Agent

100 (Trying)

100.1.0.252 100.1.0.25 100.2.0.2 100.2.0.8

INVITE1

2INVITE3

INVITE5100 (Trying)4

ACK14ACK13

200 (OK)10200 (OK)12 200 (OK)11

180 (Ringing)8180 (Ringing)9

19

18

20

17

180 (Ringing)7

6 100 (Trying)

OPTIONS15

200 (OK)16

Transaction1

Transaction2

Transaction3

Session


SIP-T(SIP for Telephone)

Basic functions of SIP-T: Encapsulating ISUP in SIP message bodies

Mapping ISUP information to SIP message headers Transfer the signaling in sessions using the INFO request


SIP Commands VS ISUP Messages

SIP MESSAGE OR RESPONSE ISUP MESSAGE

INVITE IAM or SAM

INFO USR

BYE REL

CANCEL REL

ACK ——

REGISTER ——

18x ACM or CPG

200 OK (to INVITE) ANM or CON

4xx, 5xx, 6xx REL

200 OK (to BYE) RLC


PSTN switch SIP-T gateway SIP-T gateway PSTN switch

1 IAM2 INVITE (IAM)

3 IAM4 100 Trying

5 ACM

7 ACM One-wayRTP Media One-way speechOne-way speech

8 ANM9 200 OK(ANM)

10 ANM

11 ACK

Two-way speech RTP Media Session Two-way speech

12 REL13 BYE(REL)

14 REL

15 RLC16 200 OK(RLC)17 RLC

No Speech PathNo Media Session No Speech Path

6 183 Session Progress (ACM)6 180 Ringing (ACM)

session

Transaction1

Transaction2


Agenda

H.248 Protocol Introduction SIP Protocol Introduction SigTran and M3UA / SCTP Service Flow in LMSD


IETF SIGTRAN Protocol Model

IP

SCTP

SUA

M2PAM3UA

MTP3

BSSAP

SCCP

TCAP

MAP

H.248 TUP/ISUP

M2UA

MTP3IUA

Q.931

V5UA

V5.2

Blue parts are SIGTRAN (Signaling Transport).

SS7 SCCP-User Adaptation Layer

SS7 MTP2-User Peer-to-Peer Adaptation Layer

SS7 MTP3-User Adaptation Layer

SS7 MTP2-User Adaptation Layer

ISDN Q.921-User Adaptation Layer

V5.2-User Adaptation Layer


Architecture of the SIGTRAN

Standard IP

Common signaling transfer protocol

SS7 signaling adaptation layer

{SIGTRAN

The SIGTRAN protocol system consists of two parts, signaling adaptation layer and signaling transport layer. The lower layer adopts the standard IP bearer.


Location of SIGTRAN in the Network

SCN SCNSGW SGW

MSCe MSCe

MGW MGW

RTP/RTCP MEDIAMEDIA

SIGTRAN

For H248

For handover


SCTP Protocol

• SCTP: Stream Control Transmission Protocol

• A reliable connection-oriented transport protocols

• At the same layer as TCP and UDP in network model

• SCTP is designed to transport PSTN signaling messages over IP networks

SCTP userapplication

SCTP layer

IP

SCTP userapplication

IPOne or several IP addresses

Network transportSCTP end point A SCTP end point B

SCTP layer


What does SCTP offer?

SCTPSCTP

AdvantagesAdvantages

Flow control and

Congestion avoidance

Network-level fault

tolerance

Protection against

flooding

Protection against

attacks


Why not TCP?

TCPTCP

LimitationsLimitations

Byte-stream

Oriented

Strict order of

Transmission control

No application specific

Control of protocol

parameters

Limited scope of

TCP socket

Vulnerable forDoS attack


Why not UDP?

UDPUDP

LimitationsLimitations

Unreliable No acknowledgement

No congestion controlMulti-casting adds

Unnecessary traffic

… too many


Development of SCTP

TCP SCTP

UDP

USCTP


SCTP In the SIGTRAN Model

Transport Protocol

IP

SCTP

IUA M2UA/M2PA M3UA SUA

Q931/QSIG MTP3 TUP ISUP TCAP

Signaling Adaptation Layer

IP Transport Layer

SCCP

TCAPSignaling Application Layer


Association

• An association is an SCTP connection. • The creation of an association is initiated by an SCTP user.

To guarantee the security and prevent malicious attacks, the cookie mechanism is adopted during the starting of an association.

• The connection creation and release is used for the transition of the connection status and abnormality handling.


M3UA Protocol

• M3UA provides powerful routing function, perfect signaling network management function and powerful networking capability and is applicable to inter-network telephone interconnection.


M3UA Application in SGW

MTP1

MTP2

MTP3

ISUP

MTP1

MTP2

MTP3

ISUP

SEP STP MSCe(ASP)SG(SGP)

TDM IP

IP

SCTP

M3UA

IP

SCTP

M3UA

NIF ISUP


M3UA Application between IPSPs

M3UA

SCTP

IP

IPSPIP

SCCP

SCCPUSER

M3UA

SCTP

IP

IPSP

SCCP

SCCPUSER


Conception of M3UA

• Application Server (AS)– A logic entity that serves specific routing keys. It includes a group of ap

plication server processes, one or some of which are activated to handle services.

• Application Server Process (ASP)– It is the process instance of the AS and is used as the active or standby p

rocess of the AS. ASP can include SCTP end point and can be configured to handle several AS signaling services.


Conception of M3UA.

• IP Server Process (IPSP) – It is the process instance based on IP application. IPSP is

essentially identical with ASP, but uses point-to-point M3UA instead of SG service.

• Signaling Gateway (SG)– SG receives or sends the SS7 upper-layer user messages at the

boundary between IP network and SS7 signaling network. – It is a signaling point in SS7 signaling network. It includes one or

several signaling gateway processes, one or more of which normally handles the service.

• Signaling Gateway Process (SGP)– SGP is the process instance of signaling gateway and is used as the

active, standby or load sharing process of the signaling gateway.


Conception of M3UA.

• Routing Key– A routing key describes a set of SS7 signaling parameters and

parameter values and uniquely defines a signaling service handled by a specific application server. The parameters of the routing keys can not be based on several destination signaling point codes.

– The routing keys used in M3UA include: DPC, SIO+DPC, SIO+DPC+OPC and SIO+DPC+OPC+CIC.

• Routing context– It uniquely identifies the values of the routing keys.


Agenda



3G Call 3G Service Flow3G BS OMSCe OMGW HLRe SMSCe SMGW 3G BSCM Service Request

Megaco: ADD [RTP, sendrecv, SDP-BS1]; ADD [RTP, recvonly]

OMGW-Context

Megaco: Reply [T1, T2]

T1

T2

Assignment Request

Assignment Complete

ANSI-41: LOCREQ [DGTSDIAL]

ANSI-41:ROUTREQ [MSID]

ANSI-41:routreq [TLDN]

ANSI-41: locreq [TRMLIST, REDIND]

SIP-T/SIP: INVITE [IAM, T2]

SIP-T/SIP: 183 Session Progress[ACM] /180 Ringing[ACM]

Megaco:MODIFY[Ringback]

Megaco: Reply []

Origination-Side Ringback

Paging Request

OBSC-Context

T0


3G Call 3G Service Flow _Cont.3G BS OMSCe OMGW HLRe SMSCe SMGW 3G BS

OMGW-Context

T1

T2

Paging Response

Megaco: Reply []

SMGW-Context

Megaco: ADD [RTP, Ringback, sendrecv, T2]; ADD [RTP, sendrecv, SDP-BS2]

T3

T4

Megaco: Reply [T3, T4]

SIP-T/SIP: 183 Session Progress [T3]

Megaco:MODIFY[Ringback off];MODIFY[RTP,T3,sendrecv];

Megaco: Reply []

Bearer Update Request

Bearer Update Response

Termination-Side Ringback

Assignment Request

Assignment Complete

Connect

SIP-T/SIP: 200 OK(INVITE) [ANM]

Megaco: Modify [Ringback off]

SIP-T/SIP: ACK (INVITE)

Conversation

OBSC-Context

T0

SBSC-Context

T5


Handover from 3G BS TO 3G BS

Terminating/

Anchor MSCe

Terminating/

Anchor MGW

Originating/

Anchor MSCe

Originating/

Anchor MGW

Originating

BS

zz yy

A1p A2p

39

39

Before Handoff

Terminating/

Anchor MSCe

Terminating/

Anchor MGW

Originating/

Anchor MSCe

Originating/

Anchor MGW

Originating

BS

zz yy

39

39Target/

Anchor MSCe

Target/

Anchor MGW

39

Target

BS

yy

zz

A1p A2p

After Handoff


Service Flow of Handover from 3G BS TO 3G BS

3G ABS AMSCe AMGW TMGW TMSCe 3G TBS

AMGW-Context

Tb Tc

ABSC-Context

TaHANDOFF REQUIRED

Td

H.248: ADD[C1, $, isolate]

H.248: Reply[Td]

ANSI41: FACDIR2[vCIC]

SIP: INVITE [ SDP-Td; vCIC ]

H.248: ADD[$, $, $, sendreceive]

H.248: Reply[C2, Te, Tf]

TMGW-Context

Te Tf

HANDOFF REQUEST [Tf]

HANDOFF REQUEST ACK [Tg]

TBSC-Context

Tg

H.248: MODIFY[C2, Tf, Tg]

H.248: Reply[]

SIP: 200 OK (INVITE) [SDP-Te]

H.248: MODIFY[C1, Td, Te]

H.248: Reply[]

SIP: ACK

ANSI41: facdir2

HANDOFF COMMAND

HANDOFF COMMENCED


Service Flow of Handover from 3G BS TO 3G BS _Cont.

3G ABS AMSCe AMGW TMGW TMSCe 3G TBS

BEARER UPDATE REQUEST

BEARER UPDATE RESPONSE

AMGW-Context

Tb Tc

ABSC-Context

Ta

Td

TMGW-Context

Te Tf

TBSC-Context

Tg

HANDOFF COMPLETE

ANSI41: MSONCH

H.248: MODIFY[C1, Tb, isolate, Td, bothway]

H.248: Reply[]

H.248: Subtract[Tb]

H.248: Reply[]

CLEAR COMMAND

CLEAR COMPLETE


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10 h.248,sip,sigtran and service flow of lmds

Documents

property of zte corporation

kind of disclosing

mgw mgw mgw media

termination termination

msce control

ip msce

context context

handshake msce mgw port